diff options
Diffstat (limited to 'libavcodec/ac3dec.h')
| -rw-r--r-- | libavcodec/ac3dec.h | 47 |
1 files changed, 29 insertions, 18 deletions
diff --git a/libavcodec/ac3dec.h b/libavcodec/ac3dec.h index babd0a7d70..3ac44ee17e 100644 --- a/libavcodec/ac3dec.h +++ b/libavcodec/ac3dec.h @@ -2,20 +2,20 @@ * Common code between the AC-3 and E-AC-3 decoders * Copyright (c) 2007 Bartlomiej Wolowiec <bartek.wolowiec@gmail.com> * - * This file is part of Libav. + * This file is part of FFmpeg. * - * Libav is free software; you can redistribute it and/or + * FFmpeg is free software; you can redistribute it and/or * modify it under the terms of the GNU Lesser General Public * License as published by the Free Software Foundation; either * version 2.1 of the License, or (at your option) any later version. * - * Libav is distributed in the hope that it will be useful, + * FFmpeg is distributed in the hope that it will be useful, * but WITHOUT ANY WARRANTY; without even the implied warranty of * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU * Lesser General Public License for more details. * * You should have received a copy of the GNU Lesser General Public - * License along with Libav; if not, write to the Free Software + * License along with FFmpeg; if not, write to the Free Software * Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA */ @@ -51,6 +51,7 @@ #define AVCODEC_AC3DEC_H #include "libavutil/float_dsp.h" +#include "libavutil/fixed_dsp.h" #include "libavutil/lfg.h" #include "ac3.h" #include "ac3dsp.h" @@ -97,6 +98,12 @@ typedef struct AC3DecodeContext { int dolby_headphone_mode; ///< dolby headphone mode (dheadphonmod) ///@} + int preferred_stereo_downmix; + float ltrt_center_mix_level; + float ltrt_surround_mix_level; + float loro_center_mix_level; + float loro_surround_mix_level; + ///@name Frame syntax parameters int snr_offset_strategy; ///< SNR offset strategy (snroffststr) int block_switch_syntax; ///< block switch syntax enabled (blkswe) @@ -132,8 +139,8 @@ typedef struct AC3DecodeContext { int num_spx_bands; ///< number of spx bands (nspxbnds) uint8_t spx_band_sizes[SPX_MAX_BANDS]; ///< number of bins in each spx band uint8_t first_spx_coords[AC3_MAX_CHANNELS]; ///< first spx coordinates states (firstspxcos) - float spx_noise_blend[AC3_MAX_CHANNELS][SPX_MAX_BANDS]; ///< spx noise blending factor (nblendfact) - float spx_signal_blend[AC3_MAX_CHANNELS][SPX_MAX_BANDS];///< spx signal blending factor (sblendfact) + INTFLOAT spx_noise_blend[AC3_MAX_CHANNELS][SPX_MAX_BANDS]; ///< spx noise blending factor (nblendfact) + INTFLOAT spx_signal_blend[AC3_MAX_CHANNELS][SPX_MAX_BANDS];///< spx signal blending factor (sblendfact) ///@} ///@name Adaptive hybrid transform @@ -145,15 +152,15 @@ typedef struct AC3DecodeContext { int fbw_channels; ///< number of full-bandwidth channels int channels; ///< number of total channels int lfe_ch; ///< index of LFE channel - float downmix_coeffs[AC3_MAX_CHANNELS][2]; ///< stereo downmix coefficients + SHORTFLOAT downmix_coeffs[AC3_MAX_CHANNELS][2]; ///< stereo downmix coefficients int downmixed; ///< indicates if coeffs are currently downmixed int output_mode; ///< output channel configuration int out_channels; ///< number of output channels ///@} ///@name Dynamic range - float dynamic_range[2]; ///< dynamic range - float drc_scale; ///< percentage of dynamic range compression to be applied + INTFLOAT dynamic_range[2]; ///< dynamic range + INTFLOAT drc_scale; ///< percentage of dynamic range compression to be applied ///@} ///@name Bandwidth @@ -201,22 +208,26 @@ typedef struct AC3DecodeContext { ///@name Optimization BswapDSPContext bdsp; +#if USE_FIXED + AVFixedDSPContext *fdsp; +#else AVFloatDSPContext fdsp; +#endif AC3DSPContext ac3dsp; FmtConvertContext fmt_conv; ///< optimized conversion functions ///@} - float *outptr[AC3_MAX_CHANNELS]; - float *xcfptr[AC3_MAX_CHANNELS]; - float *dlyptr[AC3_MAX_CHANNELS]; + SHORTFLOAT *outptr[AC3_MAX_CHANNELS]; + INTFLOAT *xcfptr[AC3_MAX_CHANNELS]; + INTFLOAT *dlyptr[AC3_MAX_CHANNELS]; ///@name Aligned arrays - DECLARE_ALIGNED(16, int32_t, fixed_coeffs)[AC3_MAX_CHANNELS][AC3_MAX_COEFS]; ///< fixed-point transform coefficients - DECLARE_ALIGNED(32, float, transform_coeffs)[AC3_MAX_CHANNELS][AC3_MAX_COEFS]; ///< transform coefficients - DECLARE_ALIGNED(32, float, delay)[AC3_MAX_CHANNELS][AC3_BLOCK_SIZE]; ///< delay - added to the next block - DECLARE_ALIGNED(32, float, window)[AC3_BLOCK_SIZE]; ///< window coefficients - DECLARE_ALIGNED(32, float, tmp_output)[AC3_BLOCK_SIZE]; ///< temporary storage for output before windowing - DECLARE_ALIGNED(32, float, output)[AC3_MAX_CHANNELS][AC3_BLOCK_SIZE]; ///< output after imdct transform and windowing + DECLARE_ALIGNED(16, int, fixed_coeffs)[AC3_MAX_CHANNELS][AC3_MAX_COEFS]; ///< fixed-point transform coefficients + DECLARE_ALIGNED(32, INTFLOAT, transform_coeffs)[AC3_MAX_CHANNELS][AC3_MAX_COEFS]; ///< transform coefficients + DECLARE_ALIGNED(32, INTFLOAT, delay)[AC3_MAX_CHANNELS][AC3_BLOCK_SIZE]; ///< delay - added to the next block + DECLARE_ALIGNED(32, INTFLOAT, window)[AC3_BLOCK_SIZE]; ///< window coefficients + DECLARE_ALIGNED(32, INTFLOAT, tmp_output)[AC3_BLOCK_SIZE]; ///< temporary storage for output before windowing + DECLARE_ALIGNED(32, SHORTFLOAT, output)[AC3_MAX_CHANNELS][AC3_BLOCK_SIZE]; ///< output after imdct transform and windowing DECLARE_ALIGNED(32, uint8_t, input_buffer)[AC3_FRAME_BUFFER_SIZE + FF_INPUT_BUFFER_PADDING_SIZE]; ///< temp buffer to prevent overread ///@} } AC3DecodeContext; |